As covered in Chapter 1, all circuit-switched networks today work on the premise of switching calls at the data link layer. The circuit switches are organized in a hierarchical model in which switches higher in the hierarchy are called tandem switches.
Tandem switches do not actually terminate any local loops; rather, they act as a higher-layer circuit switch. In the hierarchical model, several layers of tandem circuit switches can exist, as shown in Figure 7-6. This enables end-to-end connectivity for anyone with a phone, without the need for a direct connection between every home on the planet.
Figure 7-6 Tandem Switching Hierarchy
Typically, a voice call that passes through the two TDM switches and one tandem switch does not incur degradation in voice quality because these circuit switches use 64 Kbps channels.
If the TDM switches compress voice and the tandem switch must decompress and recompress the voice, the voice quality can be drastically affected. Although compression and recompression are not common in the PSTN today, you must plan for it and design around it in packet networks.
Voice degradation occurs when you have more than one compression/decompression cycle for each phone call. Figure 7-7 provides an example of when this scenario might occur.
Figure 7-7 VoIP Tandem Encoding
Figure 7-7 depicts three VoIP routers connected and acting as tie-lines between one central-site PBX and three remote-branch PBXs. The network is designed to put all the dial-plan information in the central-site PBX. This is common in many enterprise networks to keep the administration of the dial plan centralized.
A drawback to tandem encoding when used with VoIP is that, if a telephony user at branch B wants to call a user at branch C, two VoIP ports at central site A must be utilized. Also, two compression/ decompression cycles exist, which means that voice quality will degrade.
Different codecs react differently to tandem encoding. G.729 can handle two compression/ decompression cycles, while G.723.1 is less resilient to multiple compression cycles.
Assume, for example, that a user at remote site B wants to call a user at remote site C. The call goes through PBX B, is compressed and packetized at VoIP router B, and is sent to the central site VoIP router A, which decompresses the call and sends it to PBX A. PBX A circuit-switches the call back to its VoIP router (router A), which compresses and packetizes the call, and sends it to the remote site C, where it is then decompressed and sent to PBX C. This process is known as tandem-compression; you should avoid it in all networks where compression exists.
It is easy to avoid tandem compression. This customer simplified the router configuration at the expense of voice quality. Cisco IOS has other mechanisms that can simplify management of dial plans and still keep the highest voice quality possible.
One possible method is to use a Cisco IOS Multimedia Conference Manager (for instance, H.323 Gatekeeper). Another mechanism is to use one of Cisco's management applications, such as Cisco Voice Manager, to assist in configuring and maintaining dial plans on all your routers.
Taking the same example of three PBXs connected through three VoIP routers, but configuring the VoIP routers differently, simplifies the call-flow and avoids tandem encoding, as shown in Figure 7-8.
Figure 7-8 VoIP Without Tandem Encoding
You can see one of IP's strengths in Figure 7-8: a tie-line does not have to be leased from the telephone company to complete calls between two PBXs. If a data network connects the sites, VoIP can ride across that network.
The dial plan is moved from the central-site PBX to each of the VoIP routers. This enables each VoIP device to make a call-routing decision and removes the need for tie-lines. The major benefit of this change is the removal of needless compression/decompression cycles.